The invention relates to a method and an apparatus for sampling-rate conversion of audio signals, especially for sampling-rate conversion at the decoding of MPEG audio or DOLBY AC-3 encoded signals.
Today many different sampling rates exist like 44.1 kHz for Compact Disc, 32 kHz and 48 kHz for DAT, Digital-VCR or Sat-TV and 48 kHz or 96 kHz for DVD audio signals. Therefore, a change of the sampling-rate is necessary if the internal sampling-rate of the decoder of a playback or recording device differs from the sampling-rate of the audio signal to be decoded.
Generally, performing sampling-rate conversion from higher to lower sampling frequency (e.g. 48 to 32 kHz) results in aliasing like shown in FIG. 1. The schematic spectrum SPEC1 of the digital signal sampled with fs1 is shown in FIG. 1a). After re-sampling with fs2 less than fs1, the digital signal has a spectrum SPEC2 according to FIG. 1b). The overlapping regions OV already show that aliasing errors have occurred. After consecutive D/A conversion and appropriate low-pass filtering an analog signal with spectrum SPEC3 shown in FIG. 1c) results, which contains severe alias distortion AL.
It is known to use a low-pass filter, known as anti-alias filter, for reducing or totally avoiding this alias distortion by decreasing or removing spectral contents above fs2/2 from the digital signal. However, calculating the anti-alias filter requires additional processing power which is desirable to save.
The invention is based on the object of specifying a method for sampling-rate conversion of audio signals without the use of an anti-alias filter. This object is achieved by means of the method specified in claim 1.
The invention is based on the further object of specifying an apparatus for carrying out the method according to the invention. This object is achieved by means of the apparatus specified in claim 6.
In principle, the method for sampling-rate conversion of audio signals, which are sampled with a first sampling frequency before spectral encoding and are re-sampled after spectral decoding with a second sampling frequency which is smaller than the first sampling frequency, consists in the fact that the signal parts beyond an upper frequency limit are reduced, advantageously suppressed, at spectral decoding resulting in a bandwidth of the signal to be re-sampled which is less than half of the second sampling frequency.
This method not only totally removes the processing power needed for calculating an anti-alias filter, but also limits the decoding work needed.
In an advantageous manner, the spectral encoding algorithm uses subband coding.
In this case, it may be particularly advantageous if the subband coding algorithm corresponds to the MPEG standard and the decoding is limited to the first 20 subbands.
In a further advantageous development the spectral encoding algorithm uses a transformation into the frequency domain, e.g. DFT.
In this case, it may be particularly advantageous if the spectral encoding algorithm corresponds to the AC-3 standard and spectral lines are reduced or set to zero at decoding.
In principle, the apparatus for carrying out the inventive method consists in a sample rate converter for sampling-rate conversion of a digital audio signal from a first sampling-rate to a second sampling-rate, consists in the fact that a frequency-to-time converter suppresses the signal parts of the digital audio signal beyond an upper frequency limit resulting in a bandwidth of the signal to be re-sampled which is equal to or less than half of the second sampling frequency.
The invention may be particularly advantageous if the apparatus is part of an audio decoder for decoding audio data according to any MPEG audio or the DOLBY AC-3 standard.